asterisk users - Jan 2005 - MoH does not de-attach

Hi

We have a fairly simple Asterisk setup for a callcenter: around 10-15
operators running SIP softphones (X-Pro) and an Asterisk box connected
to a E1 service using a Digium T100P, and to our legacy PBX (NEC) over
a Digium TDM400P FXO interfaces.

Everything is working except our Music on Hold after a transfer of an
incoming call to our old PBX, that does not de-attach itself from the
transfer. For example:
1. Call from customer enters Asterisk through E1 channel
2. Call center operator pickups the call, to find out he needs to xfer
it to some extension on the old PBX, lets say ext. 300
3. He puts the customer on hold. At that moment, the customer starts
to listen to MoH.
4. He pickups another line on the SIP phone, and calls extension 300
of the older PBX, through one of the FXO interfaces, and "announces"
the call
5. Operator transfer the original call to the second line, and
de-attach itself from the call

The transfer is always successful, but sometimes (lets say bout 50% of
the times), the MusicOnHold does not "de-attach" itself from the call,
so both the customer and ext. 300 on the old PBX CONTINUES to hear the
MusicOnHold on the background, not being able to mute it. As you may
imagine, this is somewhat annoying.

We have tried with different MoH setup: initially using mpg123 player,
and now using slimserver, but it keeps happening.

Any clue whats going on?

Thanks for your advice.
Regards.

Hi!
> The transfer is always successful, but sometimes (lets say bout 50% of
> the times), the MusicOnHold does not "de-attach" itself from the
call,
Maybe a good start for an investigation is to try a different softphone 
for the operator and see if it makes a difference. Then put Asterisk into 
"sip debug" mode and see what is happening, try to find out why it 
happens at times, and why at other times it doesn't - where is the 
difference? 

This is a very strange problem, never heard of something like this before 
- which makes me believe that your NEC PBX could be part of the problem. 
Anyway, it would help if you stated which Asterisk version you are using.

Cheers, Philipp